live555关于RTSP协议交互流程

//创建RTSPServer类对象
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
//createNew实现
RTSPServer*
RTSPServer::createNew(UsageEnvironment& env, Port ourPort,UserAuthenticationDatabase* authDatabase,unsigned reclamationSeconds) {int ourSocketIPv4 = setUpOurSocket(env, ourPort, AF_INET);int ourSocketIPv6 = setUpOurSocket(env, ourPort, AF_INET6);if (ourSocketIPv4 < 0 && ourSocketIPv6 < 0) return NULL;return new RTSPServer(env, ourSocketIPv4, ourSocketIPv6, ourPort, authDatabase, reclamationSeconds);
}

//RTSPServer构造函数
RTSPServer::RTSPServer(UsageEnvironment& env,int ourSocketIPv4, int ourSocketIPv6, Port ourPort,UserAuthenticationDatabase* authDatabase,unsigned reclamationSeconds): GenericMediaServer(env, ourSocketIPv4, ourSocketIPv6, ourPort, reclamationSeconds),fHTTPServerSocketIPv4(-1), fHTTPServerSocketIPv6(-1), fHTTPServerPort(0),fClientConnectionsForHTTPTunneling(NULL), // will get created if neededfTCPStreamingDatabase(HashTable::create(ONE_WORD_HASH_KEYS)),fPendingRegisterOrDeregisterRequests(HashTable::create(ONE_WORD_HASH_KEYS)),fRegisterOrDeregisterRequestCounter(0), fAuthDB(authDatabase),fAllowStreamingRTPOverTCP(True),fOurConnectionsUseTLS(False), fWeServeSRTP(False) {
}
//GenericMediaServer构造函数
GenericMediaServer
::GenericMediaServer(UsageEnvironment& env, int ourSocketIPv4, int ourSocketIPv6, Port ourPort,unsigned reclamationSeconds): Medium(env),fServerSocketIPv4(ourSocketIPv4), fServerSocketIPv6(ourSocketIPv6),fServerPort(ourPort), fReclamationSeconds(reclamationSeconds),fServerMediaSessions(HashTable::create(STRING_HASH_KEYS)),fClientConnections(HashTable::create(ONE_WORD_HASH_KEYS)),fClientSessions(HashTable::create(STRING_HASH_KEYS)),fPreviousClientSessionId(0),fTLSCertificateFileName(NULL), fTLSPrivateKeyFileName(NULL) {ignoreSigPipeOnSocket(fServerSocketIPv4); // so that clients on the same host that are killed don't also kill usignoreSigPipeOnSocket(fServerSocketIPv6); // ditto// Arrange to handle connections from others:env.taskScheduler().turnOnBackgroundReadHandling(fServerSocketIPv4, incomingConnectionHandlerIPv4, this);env.taskScheduler().turnOnBackgroundReadHandling(fServerSocketIPv6, incomingConnectionHandlerIPv6, this);
}

void GenericMediaServer::incomingConnectionHandlerOnSocket(int serverSocket) {struct sockaddr_storage clientAddr;SOCKLEN_T clientAddrLen = sizeof clientAddr;int clientSocket = accept(serverSocket, (struct sockaddr*)&clientAddr, &clientAddrLen);if (clientSocket < 0) {int err = envir().getErrno();if (err != EWOULDBLOCK) {envir().setResultErrMsg("accept() failed: ");}return;}ignoreSigPipeOnSocket(clientSocket); // so that clients on the same host that are killed don't also kill usmakeSocketNonBlocking(clientSocket);increaseSendBufferTo(envir(), clientSocket, 50*1024);#ifdef DEBUGenvir() << "accept()ed connection from " << AddressString(clientAddr).val() << "\n";
#endif// Create a new object for handling this connection:(void)createNewClientConnection(clientSocket, clientAddr);
}
//createNewClientConnection函数实现
GenericMediaServer::ClientConnection*
RTSPServer::createNewClientConnection(int clientSocket, struct sockaddr_storage const& clientAddr) {return new RTSPClientConnection(*this, clientSocket, clientAddr, fOurConnectionsUseTLS);
}

GenericMediaServer::ClientConnection
::ClientConnection(GenericMediaServer& ourServer,int clientSocket, struct sockaddr_storage const& clientAddr,Boolean useTLS): fOurServer(ourServer), fOurSocket(clientSocket), fClientAddr(clientAddr), fTLS(envir()) {fInputTLS = fOutputTLS = &fTLS;// Add ourself to our 'client connections' table:fOurServer.fClientConnections->Add((char const*)this, this);if (useTLS) {// Perform extra processing to handle a TLS connection:fTLS.setCertificateAndPrivateKeyFileNames(ourServer.fTLSCertificateFileName,ourServer.fTLSPrivateKeyFileName);fTLS.isNeeded = True;fTLS.tlsAcceptIsNeeded = True; // call fTLS.accept() the next time the socket is readable}// Arrange to handle incoming requests:resetRequestBuffer();envir().taskScheduler().setBackgroundHandling(fOurSocket, SOCKET_READABLE|SOCKET_EXCEPTION, incomingRequestHandler, this);
}
//incomingRequestHandler函数最终调用
void GenericMediaServer::ClientConnection::incomingRequestHandler() {if (fInputTLS->tlsAcceptIsNeeded) { // we need to successfully call fInputTLS->accept() first:if (fInputTLS->accept(fOurSocket) <= 0) return; // either an error, or we need to try again laterfInputTLS->tlsAcceptIsNeeded = False;// We can now read data, as usual:}int bytesRead;if (fInputTLS->isNeeded) {bytesRead = fInputTLS->read(&fRequestBuffer[fRequestBytesAlreadySeen], fRequestBufferBytesLeft);} else {struct sockaddr_storage dummy; // 'from' address, meaningless in this casebytesRead = readSocket(envir(), fOurSocket, &fRequestBuffer[fRequestBytesAlreadySeen], fRequestBufferBytesLeft, dummy);}handleRequestBytes(bytesRead);//该函数实现了对 OPTION DESCRIBE SETUP等各种信令的处理逻辑
}

subsession->getStreamParameters(fOurSessionId, fOurClientConnection->fClientAddr,clientRTPPort, clientRTCPPort,fStreamStates[trackNum].tcpSocketNum, rtpChannelId, rtcpChannelId,&fOurClientConnection->fTLS,destinationAddress, destinationTTL, fIsMulticast,serverRTPPort, serverRTCPPort,fStreamStates[trackNum].streamToken);

void OnDemandServerMediaSubsession ::getStreamParameters(...)
{...if (clientRTPPort.num() != 0 || tcpSocketNum >= 0){ // Normal case: Create destinationsportNumBits serverPortNum;if (clientRTCPPort.num() == 0){// We're streaming raw UDP (not RTP). Create a single groupsock:NoReuse dummy(envir()); // ensures that we skip over ports that are already in usefor (serverPortNum = fInitialPortNum;; ++serverPortNum){serverRTPPort = serverPortNum;rtpGroupsock = createGroupsock(nullAddress(destinationAddress.ss_family), serverRTPPort);if (rtpGroupsock->socketNum() >= 0)break; // success}udpSink = BasicUDPSink::createNew(envir(), rtpGroupsock);}else{// Normal case: We're streaming RTP (over UDP or TCP).  Create a pair of// groupsocks (RTP and RTCP), with adjacent port numbers (RTP port number even).// (If we're multiplexing RTCP and RTP over the same port number, it can be odd or even.)NoReuse dummy(envir()); // ensures that we skip over ports that are already in usefor (portNumBits serverPortNum = fInitialPortNum;; ++serverPortNum){serverRTPPort = serverPortNum;//创建RTP端口(rtp的UDP套接字)rtpGroupsock = createGroupsock(nullAddress(destinationAddress.ss_family), serverRTPPort);if (rtpGroupsock->socketNum() < 0){delete rtpGroupsock;continue; // try again}if (fMultiplexRTCPWithRTP){// Use the RTP 'groupsock' object for RTCP as well:serverRTCPPort = serverRTPPort;rtcpGroupsock = rtpGroupsock;}else{// Create a separate 'groupsock' object (with the next (odd) port number) for RTCP://RTCP端口号在RTP端口号的基础上加1serverRTCPPort = ++serverPortNum;//创建RTCP端口(rtcp的UDP套接字)rtcpGroupsock = createGroupsock(nullAddress(destinationAddress.ss_family), serverRTCPPort);if (rtcpGroupsock->socketNum() < 0){delete rtpGroupsock;delete rtcpGroupsock;continue; // try again}}break; // success}unsigned char rtpPayloadType = 96 + trackNumber() - 1; // if dynamicrtpSink = mediaSource == NULL ? NULL: createNewRTPSink(rtpGroupsock, rtpPayloadType, mediaSource);if (rtpSink != NULL){if (fParentSession->streamingUsesSRTP){rtpSink->setupForSRTP(fMIKEYStateMessage, fMIKEYStateMessageSize);}if (rtpSink->estimatedBitrate() > 0)streamBitrate = rtpSink->estimatedBitrate();}}...
}

OnDemandServerMediaSubsession(UsageEnvironment& env, Boolean reuseFirstSource,portNumBits initialPortNum = 6970,Boolean multiplexRTCPWithRTP = False);

 for (portNumBits serverPortNum = fInitialPortNum;; ++serverPortNum){serverRTPPort = serverPortNum;rtpGroupsock = createGroupsock(nullAddress(destinationAddress.ss_family), serverRTPPort);if (rtpGroupsock->socketNum() < 0){delete rtpGroupsock;continue; // try again}...}

H264LiveVideoServerMediaSubssion::H264LiveVideoServerMediaSubssion(UsageEnvironment &env, Boolean reuseFirstSource): OnDemandServerMediaSubsession(env, reuseFirstSource, 1234) {}